Subband coded digital transmission system using some composite signals

ABSTRACT

Reproduction accuracy of, for example a digital stereo audio signal, is improved by transmitting sample data as sub-signals such as frequency subband signals. In one or more subbands, corresponding components such as left and right stereo channels are combined so that only one composite signal is transmitted per subband. An indicator signal is transmitted, indicating which subbands are combined. Scale factor signals for all subbands, and for the relative intensity of the respective subband signals which were combined, may also be transmitted. In the receiver a subband signal is derived for each channel from the composite signal, before synthesis of the full channel signals which will be reproduced.

This application is a division of U.S. patent application Ser. No.08/173,850 filed Dec. 27, 1993, which is continuation of U.S. patentapplication Ser. No. 07/997,158 filed Dec. 21, 1992, now U.S. Pat. No.5,323,396, which is a continuation of application Ser. No. 07/532,462filed on Jun. 1, 1990 by Gerardus C. P. Lokhoff for DIGITAL TRANSMISSIONSYSTEM, TRANSMITTER AND RECEIVER FOR USE IN THE TRANSMISSION SYSTEM, ANDRECORD CARRIER OBTAINED BY MEANS OF THE TRANSMITTER IN THE FORM OF ARECORDING DEVICE, now abandoned.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The invention relates to a transmission system for producing a replicaof a wideband digital signal which includes at least a first and asecond component; and more particularly to such a system which comprisesan encoder including an analyzer for altering the digital signal toobtain a number n of sub-signals for said digital signal; a transmitterfor transmitting the sub-signals for reception at a different time orplace; a receiver for receiving the sub-signals; and a decoder includinga synthesizer for combining the received sub-signals to obtainrespective replicas of the digital signal. The invention also relates toan encoding transmitter, and a decoding receiver for such a system.

2. Description of the Prior Art

One transmission system of this type is known from an article, "TheCritical Band Coder-Digital Encoding of Speech Signals Based on thePerceptual Requirement of the Auditory System" by M. E. Krasner,published in Proc. IEEE ICASSP 80, Vol. 1, pp 327-331, Apr. 9-11, 1980.This article relates to a transmission system in which the sub-signalsare signals representing frequency bands. The transmitter includes afrequency subband coding system in which the speech signal band isdivided into a plurality of subbands whose bandwidths approximate thebandwidths of the critical bands of the human ear in the respectivefrequency ranges (see FIG. 2 of this article). This division is selectedbecause, based on psycho-acoustic experiments, one can expect thatquantization noise in such a subband will be masked to an optimum extentby the signals in that subband, if during quantization allowance is madefor the noise-masking curve of the human ear. Threshold values for noisemasking by m single tones are shown in FIG. 3 of this article. Thereceiver employs a corresponding subband decoding system.

When applying frequency subband coding to a high-quality digital musicsignal, such as one according to the Compact Disc Standard which uses 16bits per signal sample at a sample frequency of 1/T=44.1 kHz, with asuitably selected bandwidth and a suitable selected quantization for therespective subbbands, the quantized output signals of the coder can berepresented by an average number of approximately 2.5 bits per signalsample. The quality of the replica of the music signal does not differperceptibly from that of the original music signal in substantially allpassages of substantially all kinds of music signals.

The subbands need not necessarily correspond to the bandwidths of thecritical bands of the human ear. For example, the subbands may haveequal bandwidths, provided that allowance is made for this indetermining the masking threshold.

The invention is also applicable to other types of transmission systems,such as those in which blocks of samples are transform coded. Suchsystems are referred to in the article "Low bit-rate coding ofhigh-quality audio signals. An introduction to the MASCAM system" by G.Thiele, G. Stoll and M. Link, published in EBU Technical Review, no.230, pp. 71-94, August 1988. In such a system the transform coefficientscorrespond to the sub-signals.

The sub-signal transmission systems described above have thedisadvantage that, in some cases, perceptible differences occur betweenthe replica and the signal which was to be transmitted. Thesedifferences are perceived as a form of distortion in the replicagenerated by the receiver. Often they are the result of the number ofbits, available for quantization of certain of the sub-signals, beingtoo low.

SUMMARY OF THE INVENTION

An object of the invention is to enable transmission of signalsrepresenting a wideband digital signal with a significant reduction ofthe distortion present in the replica generated in the receiver.

Another object of the invention is to identify sub-signals,corresponding to first and second signal components which are related toeach other, which can be combined to obtain a composite signal which istransmitted in place of those sub-signals.

Another object of the invention is to transmit digital signalscorresponding to stereo audio signals with reduced distortion in thegenerated replica.

According to the invention, a system as described above furthercomprises control circuitry for determining and optimizing bitallocation, and a signal combiner for combining selected correspondingsub-signals from the first and second components of the original digitalsignal to obtain one or more composite sub-signals, and an indicatorgenerator for generating an indicator signal indicating that thesecorresponding sub-signals are combined. The receiver is responsive tothe indicator signal, for generating a signal relating to that compositesignal and related to at least one of said first and second components.Preferably, the receiver includes a decoder which synthesizes a signalwhich is the desired replica, by combining the transmitted sub-signalsand composite sub-signals.

The invention is based on recognition that the numbers of bits madeavailable for different sub-signals are not optimally allocated, so thatquantization of certain sub-signals is too rough. This leads to audibledistortion in a replica resulting from decoding of the received signal.By selectively combining subsignals which have a correspondence orrelationship to each other, and quantizing only one compositesub-signal, so as to make more bits available for quantizing of thosesub-signals which are transmitted, the reduced quantizing distortion maymore than compensate for the slight loss of information in the replica.This is especially true when the sub-signals which are combined aresignals corresponding to a same frequency subband, such as left andright stereo, or other spatial-differentiating signals, in music oraudio transmission.

Alternatively, the composite signal may itself be quantized with agreater number of bits than if the two sub-signals were quantizedseparately.

In a preferred embodiment, a control or central processing unit, and anallocation control unit, together functioning as control or steeringcircuits, control the signal combiner to combine in each of a number m₁of said subbands the subband signals of the first and second componentsin those subbands, to obtain m₁, composite signals in said m₁ subbands,where m₁ is greater than 1. The signal generator generates an indicatorsignal identifying which subbands had their corresponding sub-signalscombined. This indicator signal will function as a steering controlsignal. The transmitter transmits these composite signals in the m₁subbands, the indicator signal, and the remaining sub-signals which havenot been combined. In this embodiment, the receiver decoder has aderiving circuit for deriving m₁ subband signals from the compositesignals in the m₁ subbands, and combining these with the subband signalswhich were transmitted.

A variation of this embodiment allows a still greater reduction of thedata. An allocation control circuit in the transmitter determines thebit availability after the m₁ subbands have been processed to form thecomposite sub-signals. If bit availability is still such thatquantization of some subbands will be too rough, then a secondevaluation is made with a greater number of subbands being combined. Forexample, in each of a number m₂ subbands the subband signals of saidsignal components are combined to obtain m₂ composite signals in said m₂subbands. The value m₂ will be greater than m₁, and will preferablyinclude all of the m₁ subbands.

In this embodiment the signal generator will then generate a differentindicator signal, identifying the m₂ subbands, and the transmittertransmits these composite signals in the m₂ subbands. In the receiverthe deriving circuit derives first the m₂ composite signals in the m₂subbands from the signal received, and then derives from these m₂composite signals, in response to the indicator signal, subband signalsin the m₂ subbands corresponding to said signal portions.

In a typical audio subband division, the m₁ subbands are the m₁ highestsubbands; and if further combining is required, then the m₂ subbands arethe m₂ highest subbands. This method of combining takes advantage of thefact that the human ear is less phase sensitive in those frequencybands. In one embodiment discussed more fully below, the value m₁ ishalf of the number M of subbands. For example, if M=32, the highest(highest frequency) 16 bands may initially be selected for combining. Avalue of 20 may be used for m₂, and the process can be repeated for m₃=24 and m₄ =28.

In yet another preferred embodiment, the transmitter comprises a scalefactor determiner, for determining a scale factor for time equivalentsignal blocks of the first and second components in the subband signals;and the transmitting section transmits these scale factors. The detectorin the receiver is adapted to detect the scale factors which have beentransmitted, and to control a multiplier for the subband signals beforethe full bandwidth signal is reconstructed in a synthesis filter.Correction for any pre-emphasis is made after reconstruction.

BRIEF DESCRIPTION OF THE DRAWING

FIG. 1 is a diagram of a second digital signal generated by atransmitter according to the invention, organized as frames eachcomposed of information packets,

FIG. 2 is a diagram of the structure of a frame according to a preferredembodiment including scale factors,

FIG. 3 is a diagram of the structure of the first portion of the frameof FIG. 2,

FIG. 4 is a block diagram of a system according to the invention

FIG. 5 is a table showing the number of information packets B in aframe, for certain values of bit rate BR and sample frequency F_(s),

FIG. 6 is a table showing the numbers of frames in a padding sequencefor different bit rates,

FIG. 7 is a table showing the system information included in the firstportion of a frame,

FIG. 8 is a table showing a distribution of information between channelsfor diffferent modes,

FIG. 9 is a table of meanings of allocation information inserted in thesecond portion of a frame,

FIGS. 10 and 11 are tables showing sequences in which allocationinformation is stored for two different formats,

FIG. 12 is a block diagram of a receiver according to the invention,

FIG. 13 is a simplified block diagram of a transmitter which records thesecond digital signal on a magnetic record carrier,

FIG. 14 is a simplified block diagram of a receiver for producing areplica signal from the magnetic record carrier of FIG. 13,

FIGS. 15a-15d are diagrams of different arrangements of scale factorsand samples in the third portion of a frame,

FIG. 16 is a block diagram of one preferred transmitter arrangement,

FIG. 17 is a diagram of another structure for the first portion of aframe,

FIG. 18 is a table showing system information included in the structureof FIG. 17,

FIG. 19 is a diagram of a structure for a portion of the structure ofFIG. 17,

FIG. 20 is a table showing bit codings in an embodiment of the structureof FIG. 19,

FIG. 21 is a table showing a sequence for allocation informationaccommodated in a second frame portion associated with the first portionof FIG. 17, for a monaural mode,

FIGS. 22a-d are tables showing sequences for allocation informationaccommodated in a second frame portion associated with the first portionof FIG. 17, for a stereo intensity mode,

FIG. 23 is a diagram of a frame structure including an additionalsignal,

FIGS. 24 is a binary number diagram relating the sample with largestabsolute value to an intermediate value used for scale factorcomputations,

FIG. 25 is a table showing quantization of scaled samples to form q-bitdigital representations, and

FIG. 26 is a table showing dequantization of the q-bit digitalrepresentations.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 shows diagrammatically the second digital signal as generated bythe transmitter and transmitted via the transmission medium. The seconddigital signal takes the form of the serial digital data stream. Thesecond digital signal comprises frames, two such frames, i.e. the framej and the frame j+1, being given in FIG. 1a. The frames, such as theframe j, comprise a plurality of information packets IP1, IP2, IP3, . .. , see FIG. 1b. Each information packet, such as IP3, comprises N bitsb₀, b₁, b₂, . . . , b_(N-1), see FIG. 1c.

Number of Packets

The number of information packets in a frame depends upon

(a) the bit rate BR with which the second digital signal is transmittedthrough the transmission medium,

(b) the number of bits N in an information packet, N being larger than1,

(c) the sample frequency F_(s) of the wide-band digital signal, and

(d) the number of samples n_(s) of the wide-band digital signal.

The information which corresponds to these packets, and which afterconversion in the transmitter belongs to the second digital signal, isincluded in one frame in the following manner. The parameter P iscomputed in conformity with the formula ##EQU1## If this computationyields an integer for P the number of information packets B in a framewill be equal to P. If the computation does not result in an integersome frames will comprise P' information packets and the other frameswill comprise P'+1 information packets. P' is the next lower integerfollowing P. The number of frames comprising P' and P'+1 informationpackets is selected in such a way that the average frame rate is equalto F_(s) /n_(s).

Hereinafter it is assumed that N=32 and n_(s) =384. The table in FIG. 5gives the number of information packets (slots) in one frame for thesevalues for N and n_(s) and for four values of the bit rate BR and threevalues for the sample frequency F_(s). It is evident that for a samplefrequency F_(s) equal to 44.1 kHz the parameter P is not an integer inall cases and that consequently a number of frames comprise 34information packets and the other frames comprise 35 information packets(when BR is 128 kbit/s). This is also illustrated in FIG. 2.

FIG. 2 shows one frame. The frame comprises P' information packets IP1,IP2 . . . IP P'. Sometimes the frame comprises P'+1 information packets.This is achieved by assigning an additional information packet (dummyslot) to the frames of P' information packets. The second column of thetable of FIG. 6 gives the number of frames in the padding sequence for asample frequency of 44.1 kHz and the aforementioned four bit rates. Thethird column specifies those frames of that number of frames in thesequence which comprise P'+1 information packets. By subtracting thenumbers in the second and the third column from each other this yieldsthe number of frames in the sequence comprising P' information packets.The (P'+1)th information packet then need not contain any information,and may then comprise for example only zeroes.

It is obvious that the bit rate BR is not necessarily limited to thefour values as given in the tables of FIGS. 5 and 6. Other (for exampleintermediate) values are also possible.

FIG. 2 shows that a frame comprises three frame portions FD1, FD2 andFD3 in this order. The first frame portion FD1 contains synchronisinginformation and system information. The second frame portion FD2contains allocation information. The third frame portion FD3 containssamples and, when applicable, scale factors of the second digitalsignal. For a further explanation it is necessary to first describe theoperation of the transmitter in the transmission system in accordancewith the invention.

The Transmission System

FIG. 4 shows diagrammatically the transmission system comprising atransmitter 1 having an input terminal 2 for receiving the wide-banddigital signal S_(BB), which may be for example a digital audio signal.In the case of an audio signal, this may be a mono signal or a stereosignal, in which case the digital signal comprises a first (leftchannel) and a second (right channel) signal component. It is assumedthat the transmitter comprises a coder for subband coding of thewide-band digital signal and that the receiver consequently comprises asubband decoder for recovering the wide-band digital signal.

The transmitter comprises an analysis filter 3 responsive to the digitalwide-band signal S_(BB) to divide the wide band into a plurality M ofsuccessive frequency subbands having band numbers m, where 1≦m≦M, whichincrease with frequency. All these subbands may have the same bandwidthbut, alternatively, the subbands may have different bandwidths. In thatcase the subbands may correspond, for example, to the bandwidths of thecritical bands of the human ear. The analysis filter generates subbandsignals S_(SB1) to S_(SBM), for the respective subbands. The transmitterfurther comprises circuits for sample-frequency reduction andblock-by-block quantization of the respective subband signals, shown asthe block 9 in FIG. 4.

Such a subband coder is known and is described, for example, in theaforementioned publications by Krasner and by Theile et al. Reference isalso made to the published European Patent Application 289,080, to whichU.S. Pat. No. 4,896,362 corresponds.

For a further description of the operation of the subband coderreference is made to these publications, which are thereforeincorporated herein by reference. Such a subband coder enables asignificant data reduction to be achieved in the signal which istransmitted to the receiver 5 through the transmission medium 4, forexample a reduction from 16 bits per sample for the wide-band digitalsignal S_(BB) to 4 bits per sample if n_(S) is 384. This means thatthere are blocks of 384 samples of the wide-band digital signal, eachsample having a length of 16 bits. If a value M=32 is assumed, thewide-band digital signal is split into 32 subband signals in theanalysis filter means 3. Now 32 (blocks of) subband signals appear onthe 32 outputs of the analysis filter means, each block comprising 12samples (the subbands have equal width) and each sample having a lengthof 16 bits. This means that on the outputs of the filter means 3 theinformation content is still equal to the information content of theblock of 384 samples of the signal S_(BB) on the input 2.

The data reduction circuit 9 operates on the output of the filter 3using the knowledge about masking. At least some of the samples in the32 blocks of 12 samples, each block for one subband, are quantised moreroughly and can thus be represented by a smaller number of bits. In thecase of a static bit allocation all the samples per subband per frameare expressed in a fixed number of bits. This number can be differentfor two or more subbands but it can also be equal for the subbands, forexample equal to 4 bits. In the case of dynamic bit allocation thenumber of bits selected for every subband may differ viewed in time, sothat sometimes even a larger data reduction can be achieved, or a higherquality with the same bit rate.

The subband signals quantised in the block 9 are applied to a generatorunit 6. Starting from the quantised subband signals this unit 6generates the second digital signal as illustrated in FIGS. 1 and 2.This second digital signal, as stated hereinbefore, can be transmitteddirectly through the medium 4. Preferably, however, this second digitalsignal is first adapted in a signal converter (not shown), such as an8-to-10 converter. Such an 8-to-10 converter is described in, forexample, European Patent Application 150,082 to which U.S. Pat. No.4,620,311 corresponds. This converter converts 8-bit data words into10-bit data words, and enables an interleaving process to be applied.De-interleaving, error correction and 10-to-8 conversion are thenperformed in the receiver.

Frame Format

The composition and content of the frames will now be explained in moredetail. The first frame portion FD1 in FIG. 2 is shown in greater detailin FIG. 3. FIG. 3 shows that the first frame portion comprises exactly32 bits and is therefore exactly equal to one information packet, namelythe first information packet IP1 of the frame. The first 16 bits of theinformation packet form the synchronising signal (or synchronisingword), and may comprise for example only "ones" The bits 16 to 31 aresystem information. The bits 16 to 23 represent the number ofinformation packets in a frame. This nu consequently corresponds to P',both for the frame comprising P' information packets and for framescomprising the additional information packet IP P'+1. P' can be at themost 254(1111 1110 in bit notation) in order to avoid resemblance to thesynchronising signal. The bits 24 to 31 provide frame formatinformation.

FIG. 7 gives an example of the arrangement and significance of thisinformation. Bit 24 indicates the type of frame. In the case of format Athe second frame portion has another length (a different number ofinformation packets) than in the case of format B. As will becomeapparent hereinafter, the second frame portion FD2 in the A formatcomprises 8 information packets, namely the information packets IP2 toIP9 inclusive; and in the B format it comprises 4 information packets,namely the information packets IP2 to IP5 inclusive. The bits 25 and 26indicate whether copying of the information is allowed. The bits 27 to31 indicate the function mode. This means:

a) the channel mode, which indicates the type of wide-band signal (asstated hereinbefore this may be a stereo audio signal, a mono audiosignal, or an audio signal comprising two different signal componentsfor example representing the same text but in two different languages).FIG. 8 represents the channel mode. It illustrates how the signalcomponents are divided between the two channels (channel I and channelII) in the aforementioned cases.

b) the sample frequency F_(s) of the wide-band signal.

c) the emphasis which may be applied to the wide-band digital signal inthe transmitter. The values 50 and 15 μs are the time constants of theemphasis and CCITT J. The value 17 indicates a specific emphasis

standard as defined by the CCITT (Comite Consultative Internationale deTelegraphie et Telephonie).

The content of the frame portion FD2 in FIG. 2 will be described in moredetail with reference to FIGS. 9, 10 and 11. In the A format the secondframe portion contains eight information packets. This is based on theassumptions that the wide-band digital signal S_(BB) is converted into32 subband signals (for every signal portion of the digital signalS_(BB)), and that an allocation word having a length of four bits isassigned to every subband. This yields a total of 64 allocation wordshaving a length of 4 bits each, which can be accommodated exactly ineight information packets. In the B format the second frame portionaccommodates the allocation information for only half the number ofsubbands, so that now the second frame portion comprises only 4information packets.

FIG. 9 illustrates the significance of the four-bit allocation words AW.An allocation word associated with a specific subband specifies thenumber of bits by which the samples of the subband signal in therelevant subband are represented after quantisation in the unit 9. Forexample, the allocation word AW which is 0100 indicates that the samplesare represented by 5-bit words. Moreover, it follows from FIG. 9 thatthe allocation word 0000 indicates that no samples have been generatedin the relevant subband. This may happen, for example, if the subbandsignal in an adjacent subband has such a large amplitude that thissignal fully masks the subband signal in the relevant subband. Theallocation word 1111 is not used because it closely resembles the syncword in the first information packet IP1.

FIG. 10 indicates the sequence, in the case that the frame mode is A, inwhich the allocation words AW, j,m associated with the two channels j,where j=I or II, and the 32 subbands of the sequence number m, m rangingfrom 1 to 32, are arranged in the second frame portion. The allocationword AWl,1 belonging to the first subband signal component of the firstand lowest subband (channel I, subband 1) is inserted first. After thisthe allocation word AWII,1 belonging to the second subband-signalcomponent of the first and lowest subband (channel II, subband 1) isinserted in the second frame portion FD2. Subsequently, the allocationword AWl,2 belonging to the first subband-signal component of the secondand lowest but one subband (channel I, subband 2) is inserted in theframe portion FD2. This is followed by the allocation word AW II,2belonging to the second subband-signal component of the second subband(channel II, subband 2). This sequence continues until the allocationword AW II,4 belonging to the second subband-signal component of thefourth subband (channel II, subband 4) is inserted in the second frameportion FD2. The second information packet IP2 (slot 2) of the frame,which is the first information packet in the frame portion FD2 of theframe, is then filled exactly. Subsequently, the information packet IP3(slot 3) is filled with AW I,5; AW II,5; . . . AW II,8. This continuesin the sequence as illustrated in FIG. 10.

FIG. 10 merely gives the indices j-m of the inserted allocation word AW,j,m. FIG. 11 indicates the sequence for the allocation words in the caseof a B-format frame. In this case only allocation words of the subbands1 to 16 are inserted. The sequence, as is illustrated in FIG. 10,corresponds to the sequence in which the separate samples belonging to achannel j and a subband m are applied to the synthesis filter means uponreception in the receiver. This will be explained in greater detailhereinafter.

The serial data stream contains for example only frames in conformitywith the A format. In the receiver the allocation information in eachframe is then employed for correctly deriving the samples from theinformation in the third frame portion of said frame. The serial datastream may also comprise, more or less alternately, both frames inconformity with the A format and frames in conformity with the B format.However, the frames in conformity with both formats may contain samplesfor all channels and all subbands in the third frame portion. A frame inconformity with the B format then lacks in fact the allocationinformation required to derive the samples for the channels I or II ofthe subbands 17 to 32 from the third frame portion of a B format frame.

The receiver comprises a memory in which the allocation informationincluded in the second frame portion of an A format frame can be stored.If the next frame is a B format frame only the allocation informationfor the subbands 1 to 16 and the channels I and II in the memory isreplaced by the allocation information included in the second frameportion of the B format frame. The samples for the subbands 17 to 32from the third frame portion of the B format frame are derived from theallocation information for these subbands derived from the preceding Aformat frame and still present in the memory. The reason for thealternate use of A format frames and B format frames is that for somesubbands the allocation information (in the present case the allocationinformation for the higher subbands 17 to 32) does not change rapidly.Since during quantization the allocation information for the varioussubbands is available in the transmitter, this transmitter can decide togenerate a B format frame instead of an A format frame if the allocationinformation for the subbands 17 to 32 inclusive does not change(significantly). Moreover, this illustrates that now additional spacebecomes available for the inclusion of samples in the third frameportion FD3.

For a specific value of P' the third frame portion of a B format frameis four information packets longer than the third frame portion of an Aformat frame. This enables the number of bits by which the samples inthe lower subbands 1 to 16 are represented to be increased, so that forthese subbands a higher transmission accuracy can be achieved. Moreover,if it is required to quantize the lower subbands more accurately thetransmitter can automatically opt for the generation of B format frames.This may then be at the expense of the accuracy with which the highersubbands are quantized.

The third frame portion FD3 in FIG. 2 contains the samples of thequantised subband-signal components for the two channels. If theallocation word 0000 is not present in the frame portion FD2 for any ofthe subband channels this means that in the present example twelvesamples are inserted in the third frame portion FD3 for each of the 32subbands and 2 channels. This means that there are 768 samples in total.

Scale Factors

In the transmitter the samples may be multiplied by a scale factor priorto their quantization. For each of the subbands and channels theamplitudes of the twelve samples are divided by the amplitude of thatsample of the twelve samples which has the largest amplitude. In thatcase a scale factor should be transmitted for every subband and everychannel in order to enable the inverse operation to be performed uponthe samples at the receiving end. For this purpose the third frameportion then contains scale factors SF j,m, one for each of thequantised subband-signal components in the various subbands.

In the present example, scale factors are represented by 6-bit numbers,the most significant bit first, the values ranging from 000000 to111110. The scale factors of the subbands to which these are allocated,i.e. whose allocation information is non-zero, are accommodated in theleading part of the frame portion FD3 before the samples. This meansthat the scale factors are transmitted before the transmission of thesamples begins. As a result rapid decoding in the receiver 5 can beachieved without the necessity of storing all the samples in thereceiver, as will become apparent hereinafter. A scale factor SF j,m canthus represent the value by which the samples of the signal in the j-thchannel of the m-th subband have been multiplied. Conversely, the numberone divided by this value may be stored as the scale factor so that atthe receiving end it is not necessary to divide the scale factors beforethe samples are scaled up to the correct values.

For the frame format A the maximum number of scale factors is 64. If theallocation word AW j,m for a specific channel j and a specific subband mhas the value 0000, which means that for this channel and this subbandno samples are present in the frame portion FD3, it will not benecessary to include a scale factor for this channel and this subband.The number of scale factors is then smaller than 64. The sequence inwhich the scale factors SF j,m are inserted in the third frame portionFD3 is the same as that in which the allocation words have been insertedin the second frame portion. The sequence is therefore as follows: SF I,1; SF II, 1; SF I,2; SF II,2; SF I,3; SF II,3; . . . SF I,32; SF II,32.

If it is not necessary to insert a scale factor the sequence will not becomplete. The sequence may then be for example:

. . SF I,4; SF I,5; SF II,5; SF II,6; . . . .

In this case the scale factors for the fourth subband of channel II andthe sixth subband of channel I are not inserted. If the frame is a Bformat frame it may still be considered to insert scale factors in thethird frame portion for all the subbands and all the channels. However,this is not the only option. In this case it would also be possible toinsert scale factors in the third frame portion of the frame for thesubbands 1 to 16 only. In the receiver this requires a memory in whichall scale factors can be stored at the instant at which a previouslyarriving A format frame is received. Subsequently upon reception of theB format frame only the scale factors for the subbands 1 to 16 arereplaced by the scale factors included in the B format frame. The scalefactors of the previously received A format frame for the subbands 17 to32 are then used in order to restore the samples for these subbandsincluded in the third frame portion of the B format frame to the correctscale.

The samples are inserted in the third frame portion FD3 in the samesequence as the allocation words and the scale factors, one sample forevery subband of every channel in succession. According to thissequence, first all the first samples for the quantised subband signalsfor all the subbands of both channels are inserted, then all the secondsamples, . . . etc. The binary representation of the samples isarbitrary, the binary word comprising only "ones" preferably not beingused again.

The second digital signal generated by the transmitter 1 is subsequentlyapplied to the transmission medium 4 by the output 7, and by means ofthe transmission medium 4 this signal is transferred to the receiver 5.Transmission through the transmission medium 4 may be a wirelesstransmission, such as for example a radio transmission channel. Manyother transmission media are also possible. In this respect opticaltransmission may be envisaged, for example over optical fibres oroptical record carriers, such as Compact-Disc-like media, ortransmission by means of magnetic record carriers utilising RDAT orSDAT-like recording and reproducing technologies, for which reference ismade to the book "The art of digital audio" by J. Watkinson, FocalPress, London 1988.

The Receiver

As shown in FIG. 4, the receiver 5 comprises a decoder, which decodesthe signal encoded in the coder 6 of the transmitter 1 and converts itinto a replica of the wide-band digital signal supplied to the output 8.The essential information in the incoming signal is contained in thescale factors and the samples. The remainder of the information in thesecond digital signal is merely required for a "correct bookkeeping", toallow a correct decoding. The receiver first derives the synchronisingand system information from the frames. The decoding process is thenrepeated for every incoming frame.

FIG. 12 shows a more detailed version of the receiver 5 of FIG. 4. Thecoded signal (the second digital signal) is applied to a unit 11 throughthe terminal 10. For every frame, the unit 19 first detects the syncwords situated in the first 16 bits of the first frame portion. Sincethe sync words of successive frames are each time spaced apart by anintegral multiple of P' or P'+1 information packets, the sync words canbe detected very accurately. Once the receiver is in synchronism thesync word can be detected in the unit 19. To accomplish this, a timewindow having, for example, a length of one information packet is openedafter each occurrence of P' information packets, so that only that partof the incoming information is applied to the sync word detector in theunit 19. If the sync word is not detected the time window remains openfor the duration of another information packet because the precedingframe may be a frame comprising P'+1 information packets. From thesesync words a PLL in the unit 19 can derive a clock signal to control thecentral processing unit 18.

It is evident from the above that the receiver should know how manyinformation packets are contained in one frame. For this purpose theswitching means 15 are then in the upper position shown, to apply thesystem information to the processing unit 18. The system information cannow be stored in a memory 18a of the processing unit 18. The informationrelating to the number of information packets in a frame can be appliedto the unit 19 over a control-signal line 20, to open the time window atthe correct instants for sync-word detection. When the systeminformation is received the switch 15 is changed over to the lowerposition. The allocation information in the second frame portion of theframe can now be stored in the memory 18b.

If the allocation information in the incoming frame does not comprise anallocation word for all the subbands and channels this will have becomeapparent already from the detected system information. This may be forexample the information indicating whether the frame is an A-format or aB-format frame. Thus, under the influence of the relevant informationcontained in the system information, the processing unit 18 will storethe received allocation words at the correct location in the allocationmemory 18b.

It is obvious that in the present example the allocation memory 18bcomprises 64 storage positions. If no scale factors are transmitted, theelements bearing the reference numerals 11, 12 and 17 may be dispensedwith, and the content of the third frame portion of a frame is applieddirectly by the connection 16 from the input 10 to a synthesis filter21. The samples are applied to the filter 21 in the same sequence as theorder in which the filter 21 processes the samples in order toreconstruct the wide-band signal. The allocation information stored inthe memory 18b is required in order to divide the serial data stream ofsamples into individual samples in the filter 21, each sample having thecorrect number of bits. For this purpose the allocation information isapplied to the filter 21 over the line 22.

The receiver further comprises a deemphasis unit 23 which subjects thereconstructed digital signal supplied by the filter 21 to deemphasis.For a correct deemphasis the relevant information in the bits 24 to 31of the first frame portion should be applied from the memory 18a to thedeemphasis unit 23 over the line 24.

If the system uses scale factors in this format, the receiver willinclude the switch 11, the memory 12, and the multiplier 17, and thethird frame portion will contain the scale factors SF j,m. Because of acontrol signal applied by the processing unit 18 over the line 13, theswitch 11 is in the lower position at the instant at which the thirdframe portion FD3 of a frame arrives. Address signals are supplied tothe memory 12 by the processing unit 18 over the line 14. The scalefactors are then stored in the memory 12, which has 64 locations for thestorage of the 64 scale factors. If a B-format frame is being received,the processing unit 18 applies such address signals to the memory 12that only the scale factors for the subbands 1 to 16 are overwritten bythe scale factors in the B-format frame. Subsequently, as a result ofanother control signal applied over the line 13, the switch 11 ischanged to the upper position shown in the drawing, so that the samplesare applied to the multiplier 17. Using the allocation information,which is now applied to the multiplier 17 over the line 22, themultiplier first derives the individual samples of the correct bitlength from the serial data stream applied over the line 16.

The samples are then multiplied so as to restore them to the correctvalues which the original samples had prior to scaling down in thetransmitter. If the scale factors stored in the memory 12 are the scalefactor values by which the samples have been scaled down in thetransmitter, these values should first be inverted (one divided by thevalue) before application to the multiplier 17. Obviously, it is alsopossible to invert the scale factors upon reception before they arestored in the memory 12. If the scale factors in the frames are alreadyequal to the value by which the samples should be scaled up duringreception they can be stored directly in the memory 12, and can then beapplied directly to the multiplier 17.

It is evident that no memory is required to store all these samplesbefore the signal processing performed upon the samples contained in theframe begins. At the instant at which a sample arrives over the line 16all the information required for processing this sample is alreadyavailable, so that processing can be carried out immediately. Thisentire process is controlled and synchronized by control signals andclock signals applied to all the parts of the transmitter by theprocessing unit 18.

Not all the control signals are shown. This is not necessary because thedetails of operation of the receiver will be obvious to those skilled inthe art. Under control of the processing unit 18 the multiplier 17multiplies the samples by the appropriate multiplication factors. Thesamples, which have now been restored to the correct amplitude; areapplied to the reconstruction filter 18 in which the subband signals arereconverted to form the wide-band digital signal. Further description ofthe receiver is not necessary because such receivers are generallyknown, for example as described in the Thiele et al article cited above.Moreover, it will be evident that if the system information is alsotransmitted the receiver can be highly flexible and can correctly decodethe signals even if the second digital signals contain different systeminformation.

Other embodiments

FIG. 13 shows diagrammatically another embodiment of the transmitter, inthe form of a recording device for recording the wide-band digitalsignal on a record carriersuch as a magnetic record carrier 25. Theencoder 6 supplies the second digital signal to a recording device 27comprising a write head 26 by means of which the signal is recorded in atrack on the record carrier. It is then possible to record the seconddigital signal in a single track on the record carrier, for example bymeans of a helical-scan recorder. In this case the single track can bedivided into juxtaposed tracks which are inclined relative to thelongitudinal direction of the record carrier. An example of this is anRDAT-like recording method. Another method is to split the informationand simultaneously recording the split information in a plurality ofjuxtaposed tracks which extend on the record carrier in the longitudinaldirection of the record carrier. For this the use of an SDAT-likerecording method may be considered. A comprehensive description of thetwo above methods can be found in the aforementioned book "The art of adigital audio" by J. Watkinson.

Again it is to be noted that the signal supplied by the unit 6 may befirst be encoded in a signal converter. This encoding may, for example,be an 8-to-10 conversion followed by an interleaving process, asdescribed with reference to FIG. 4. If the encoded information isrecorded on the record carrier in a plurality of adjacent paralleltrack, the signal converter should also be capable of assigning theencoded information to the various tracks.

FIG. 14 shows diagrammatically an embodiment of the receiver 5, whichmay be used in conjunction with the transmitter of FIG. 13; the two mayform one apparatus which then provides transmission over time instead ofdistance. The receiver shown is a player or read device for reading therecord carrier 25 on which the wide-band digital signal has beenrecorded by means of the device shown in FIG. 13, in the form of thesecond digital signal described above. The second digital signal is readfrom a track on the record carrier by the read head 29 and is applied tothe receiver 5, which may be for example of a construction as shown inFIG. 12. Again the read device 28 may be constructed to carry out anRDAT-like or an SDAT-like reproducing method. Both methods are againdescribed comprehensively in the aforementioned book by Watkinson.

If the signal supplied by the unit 6 in the recording device shown inFIG. 13 has been converted, for example in an 8-to-10 conversion and inan interleaving step, the encoded signal read from the record carrier 25should first be de-interleaved and should be subjected to 10-to-8conversion. Moreover, if the encoded signal has been recorded in aplurality of parallel tracks the reproducing unit shown in FIG. 14should arrange the information read from these tracks in the correctsequence before further processing is applied.

FIGS. 15a-d show a number of other possibilities of inserting the scalefactors and the samples in the third frame portion FD3 of a frame. FIG.15a illustrates the above described method in which the scale factors SFfor all the subbands m and channels (I or II) are inserted in the thirdframe portion before the samples. FIG. 15b illustrates the samesituation as FIG. 15a, but in this case it diagrammatically representsthe storage capacity for the scale factors SF I,m and SF II,m and theassociated x samples for these two channels in the subband m. FIG. 15bshows the samples for the two channels in the subband m combined toblocks, whereas normally they are distributed within the third frameportion. The samples have a length of y bits. In the above example x is12 and y is now taken to be 8.

Stereo coding

FIG. 15c shows another format. The two scale factors for the first andthe second channel in the subband are still present in the third frameportion. However, instead of the x samples for both channels (the leftand right channels for a stereo signal) in the subband m (i.e. 2xsamples in total) only x samples for the subband m are included in thethird frame portion. These x samples are obtained, for example, byadding corresponding samples in each of the two channels to one another.Thus a monophonic signal is generated and transmitted for this subbandm.

The x samples in FIG. 15c each have a length of z bits. If z is equal toy this saves room in the third frame portion, which can be used forsamples requiring a more accurate quantisation. It is alternativelypossible to express the x samples of the mono signal in Z=2y (=16) bits.Such a signal processing is applied if the phase difference between theleft-hand and the right-hand signal component in a subband is irrelevantbut the waveform of the monophonic signal is important. This applies inparticular to signals in higher subbands because the phase-sensitivityof the ear for the frequency in these subbands is smaller. By expressingthe x samples of the mono signal in 16 bits the waveform is quantisedmore accurately, while the room occupied by these samples in the thirdframe portion is equal to that in the example illustrated in FIG. 15b.

Yet another possibility is to represent the samples by an intermediatenumber of bits, for example 12 bits. The signal definition is then moreaccurate than in the example illustrated in FIG. 15b, while at the sametime room is saved in the third frame portion so that the bits saved canbe allocated where the need is greater.

When the signals included in the third frame portion as illustrated inFIG. 15c are reproduced at the receiving end, a stereo effect isobtained which is referred to as "intensity stereo". Here, only theintensities of the left-channel and the right-channel signals (in thesubband m) can differ because of a different value for the scale factorsSF I,m and SF II,m. Thus different kinds of information relating to thestereo nature of the audio signal can be represented by the compositesignals and other signals which are transmitted.

FIG. 15d shows still another possibility. In this case there is only onescale factor SFm for both signal components in the subband m. This is asituation which is particularly apt to occur in low-frequency subbands.

Yet another possibility, which is not shown, is that the x samples forthe channels I and II of the subband m, as in FIG. 15b, do not haveassociated scale factors SF I,m and SF II,m. Consequently, these scalefactors are not inserted in the same third frame portion. In this casethe scale factors SF I,m and SF II,m included in the third frame portionof a preceding frame must be used for scaling up the samples in thereceiver.

All the possibilities described with reference to FIGS. 15a-d can beemployed in the transmitter in order to achieve a most efficient datatransfer over the transmission medium. Thus, frames as described withreference to different ones of FIGS. 15a-d may occur alternately in thedata stream. It will be appreciated that, if the receiver is to becapable of correctly decoding these different frames, information aboutthe structure of these frames must be included somewhere, such as in thesystem information.

The transmitter

FIG. 16 shows the transmitter in more detail, particularly with respectto combination of the various items of information to form the serialdata stream shown in FIGS. 1, 2 and 3. FIG. 16 in fact shows a moredetailed version of the encoder 6 in the transmitter 1. The encoder hasa central processing unit 30, which controls a number of the encodercircuits; and also includes a generator 31 for generating thesynchronising information and the system information described withreference to FIG.3, a generator 32 for supplying allocation information,a generator 33 (optional) for supplying the scale factors, a generator34 for supplying the samples for a frame, and a generator 35 forgenerating the additional information packet IP P'1.

The outputs of these generators are coupled to associated inputs of amultiplexer 40 shown as a five-position switch whose output is coupledto the output 7 of the encoder 6. The CPU 30 controls the multiplexer orswitch 40 over the line 53, and the various generators over the lines41.1 to 41.4.

The operation of the transmitter will be described for a mono signaldivided into M subband signals. These M subband signals S_(SB1) toS_(SBM) are applied to the encoder input terminals 45.1, 45.2, . . . ,45.M. If scale factors are to be used, blocks of samples of each of thesubband signals are processed together in the optional subband scalingunits 46.1 to 46.M. A number, for example twelve, of samples in a blockare scaled to the amplitude of the largest sample in the block. The Mscale factors are supplied to the unit 33 (if present) over the lines47.1 to 47.M. The subband signals are supplied both to an allocationcontrol unit 49 and (scaled if that option is in use) to M quantisers48.1 to 48.M. For every subband the allocation control unit 49 definesthe number of bits with which the relevant subband signal should bequantised. This allocation information is applied to the respectivequantisers 48.1 to 48.M over the lines 50.1 to 50.M, so that thesequantisers correctly quantise the 12 samples of each of the subbandsignals; and is also supplied to the generator 32. The quantized samplesof the subband signals are supplied to the generator 34 over the lines51.1 to 51.M. The generators 32, 33 and 34 arrange the allocationinformation, the scale factors and the samples in the correct sequencedescribed above.

In the position of the multiplexer or switch 40 shown, the synchronisingand system information associated with the frame to be generated issupplied by the generator 31 in the CPU 30 and fed to the encoder output7. Subsequently, the multiplexer or switch 40 responds to a controlsignal supplied by the CPU 30 over the line 53, and is set to the secondposition from the top so that the output of the generator 32 is coupledto the output 7. The sequence of allocation information as describedwith reference to FIG. 10 or 11 is the supplied. After this the switch40 is set to the third position from the top, coupling the output of thegenerator 33 to the output 7, and the generator 33 now supplies thescale factors in the correct sequence. The switch 40 is then set to thenext position, so that the output of the generator 34 is coupled to theoutput 7, and the generator 34 supplies the samples in the varioussubbands in the correct sequence. In this cycle exactly one frame isapplied to the output 7. Subsequently, the switch 40 is reset to the topposition. A new cycle is then started, in which a subsequent block of 12samples for each subband is encoded and a subsequent frame can begenerated on the output 7.

In some cases, for example if the sample frequency F_(s) is 44.1 kHz(see FIG. 5) an additional information packet (the dummy slot, see FIG.2) must be added. In that case, after the generator 34 has finishedsupplying the samples, the multiplexer or switch 40 will be set to thebottom position. The output of the generator 35 is now coupled to theoutput 7, and the generator 35 generates the additional informationpacket IP P'+1. After this the switch 40 is reset to the top position tostart the next cycle.

It will be clear that, if the signal received by the transmitter is tobe corrected for errors caused during transmission of the signal, anappropriate error coding and/or interleaving should be applied to thesecond digital signal addition, prior to transmission some modulation isusually required. Thus, the digital signal transmitted through thetransmission medium may not be directly identifiable as the secondsignal, but will be a signal which has been derived therefrom.

It will to be noted that, for example in the case that the subbands havedifferent widths, the number of samples for the various subbandsinserted in one third frame portion may differ and are likely to differ.If it is assumed, for example, that a division into three subbands isused, including a lower subband SB₁, a central subband SB₂ and an uppersubband SB₃, the upper subband may have a bandwidth which is, forexample, twice as large as that of the other two subbands. This meansthat the number of samples inserted in the third frame portion for thesubband SB₃ is probably also twice as large as for each of the othersubbands. The sequence in which the samples are applied to thereconstruction filter in the receiver may then be: the first sample ofSB₁, the first sample of SB₃, the first sample of SB₂, the second sampleof SB₃, the second sample of SB₁, the third sample of SB₃, the secondsample of SB₂, the fourth sample of SB₃, . . . etc. The sequence inwhich the allocation information for these subbands is then inserted inthe second frame portion will then be: first the allocation word forSB₁, then the allocation word of SB₃, and subsequently the allocationword for SB₂. The same applies to the scale factors. Moreover, thereceiver can derive, from the transmitted system information, that inthis case the cycle comprises groups of four samples each, each groupcomprising one sample of SB₁, one sample of SB₃, one sample of SB₂ andsubsequently another sample of SB₃.

Other Frame Arrangements

FIG. 17 shows another structure of the first frame portion FD1. Againthe first frame portion FD1 contains exactly 32 bits and thereforecorresponds to one information packet. The first 16 bits againconstitute the synchronising signal (or synchronisation word). Thesynchronisation word may also be the same as the synchronisation word ofthe first frame portion FD1 in FIG. 3, but the information accommodatedin bits 16 through 31 differs from the information in bits 16 through 31in FIG. 3. The bits b₁₆ through b₁₉ represent a 4-bit bit rate index (BRindex) number whose meaning is illustrated in the Table in FIG. 18. Ifthe bit rate index is equal to the 4-bit digital number `0000` thisdenotes the free-format condition, which means specified and that thedecoder has to depend upon the synchronisation word alone to detect thebeginning of a new frame. The 4-bit digital number `1111` is notemployed in order not to disturb the synchronisation word detection. Inthe second column of the table of FIG. 18 the bit rate index isrepresented as a decimal number corresponding to the 4-bit digitalnumber. The corresponding bit rate values are given in column 1.

With this format the first frame portion contains information related tothe number of information packets in the frame. As shown in FIG. 18, thesample frequency F_(s) is defined by one of the four possible 2-bitdigital numbers for the bits b₂₀ and b₂₁ having the values listed. Bit22 indicates whether the frame comprises a dummy slot, in which case b₂₂=`1`, or does not comprise a dummy slot, in which case b₂₂ =`0`. Alongwith other predetermined information, then, the information in the bitsb₁₆ through b₂₂ makes it possible to determine how many informationpackets are actually present in the frame.

From the number of samples of the wide-band signal whose correspondinginformation belonging to the second digital signal is accommodated inone frame, in the present example n_(s) =384, it is possible todetermine how many information packets B are present in the frame bymeans of the data in the Table in FIG. 8, the padding bit b₂₂ and theformula ##EQU2##

The bit b₂₃ is intended for specifying a future extension of the system.

This future extension will be described hereinafter. For the time beingthis bit is assumed to be `0`.

Indicator Signals

Various indicator and control signals are provided by the bits b₂₄through b₃₁, which will be described with reference to FIGS. 19 and 20.The bits b₂₄ and b₂₅ give the mode indication for the audio signal. Forthe four possibilities of this two-bit digital number FIG. 20 showswhether the wide-band digital signal is a stereo audio signal (`00`), amono signal (`11`), a bilingual signal (`10`), or an intensity stereoaudio signal (`01`). In the last-mentioned case the bits 26 and 27indicate which subbands have been processed in accordance with theintensity stereo method. In this example the respective two-bit numbers"00`, `01`, `10`, and `11` mean, respectively, that the subbands 5-32,9-32, 13-32 and 17-32 have been processed in accordance with theintensity stereo method. As stated hereinbefore intensity stereo can beapplied to the higher subbands because the ear is less phase-sensitivefor the frequencies in these subbands. The bit b₂₈ can be used as acopyright bit. If this bit is `1` this means that the information iscopy-protected and should/cannot be copied. The bit b₂₉ can indicatethat the information is original information (b₂₉ =`1`), for example inthe case of the prerecorded tapes, or information which has been copied(b₂₉ =`0`). The bits b₃₀ and b₃₁ specify the emphasis which may havebeen applied to the wide-band signal in the transmitter, for example asdescribed with reference to FIG. 7.

OVarious configurations of the second frame portion FD2 may be describedby the various mode indications represented by the bits b₂₄ through b₂₇in the first frame portion. The second frame portion comprises the 4-bitallocation words whose meaning has been described with reference to FIG.9. For the stereo mode (b₂₄, b₂₅ =00) and the bilingual mode (b₂₄, b₂₅=10) the second frame portion FD2 again has a length of 8 informationpackets (slots) and is composed as described with reference to FIG. 10.In the stereo mode `I` in FIG. 10 then represents, for example, theleft-channel component and `II` the right channel component. For thebilingual mode `I` denotes one language and `II` denotes the otherlanguage. For the mono mode (b₂₄, b₂₅ =11) the length of the secondframe portion FD2 is of course only 4 information packets (slots).

FIG. 21 illustrates the sequence of the allocation words for the varioussubbands 1 through 32 in the four information packets (slots) 2 through5. Thus, every quantity M-i represents a four-bit allocation word whichspecifies the number of bits in every sample in the subband of thesequence number i,i ranging from 1 to 32. In the intensity stereo mode(b₂₄, b₂₅ =01) there are four possibilities indicated by means of thebits b₂₆ and b₂₇, see FIG. 20. All these possibilities result in adifferent content of the second frame portion FD2.

FIGS. 22a to 22d illustrate the four different contents of the secondframe portion. If the switch bits b26, b₂₇ are `00` the signals in thesubbands 1 through 4 are normal stereo signals and the signals in thesubbands 5 through 32 are intensity-stereo signals. This means that forthe subbands 1 through 4 for the left-hand and right-hand channelcomponents in these subbands the associated allocation words should bestored in the second frame portion. In FIG. 22a this is represented bythe consecutive allocation words AW (I, 1 ); AW (R, 1 ); AW (I, 2); AW(R, 2); . . . AW (R, 4), stored in the slot 2 of the frame, i.e. thefirst slot of the second frame portion. FIG. 22a only gives the indices(i-j) of the allocation words, i being equal to L or R and indicatingthe left-hand and the right-hand channel component respectively, and jranging from 1 through 4 and representing the sequence number of thesubband. For the subbands 5 through 32 the left-hand and the right-handchannel components contain the same series of samples. The onlydifference resides in the scale factors for the left-hand and theright-hand channel components in a subband. Consequently, such a subbandrequires only one allocation word. The allocation words AW (i, j) forthese subbands 5 through 32 are indicated by the indices M-j, where i isconsequently equal to M for all the subbands and where j ranges from 5through 32.

FIG. 22a shows that 41/2 information packets are required for insertingthe 36 allocation words in the second frame portion. If the switch bitsb₂₆, b₂₇ are `01`, the signals in the subbands 1 through 8 will benormal stereo signals and the signals in the subbands 9 through 32 willbe intensity-stereo signals. This means that for each of the subbands 1through 8 two allocation words AW(L, j) and AW(R,j) are required andthat for each of the subbands 9 through 32 only one allocation wordAW(M,j) is required. This implies that in total 40 allocation words areneeded, included in five information packets (slots), i.e. IP2 throughIP6, of the frame. This is illustrated in FIG. 22b. In this case thesecond frame portion FD2 has a length of five information packets(slots).

If the switch bits b₂₆, b₂₇ are `10` the signals in the subbands 1through 12 will be normal stereo signals and the signals in the subbands13 through 32 will be intensity-stereo signals. FIG. 22c gives thestructure of the second frame portion FD2 with the allocation words forthe various subbands. The second frame portion now has a length of 51/2information packets (slots) in order to accommodate all the allocationwords. If the switch bits b₂₆, b₂₇ are `11` the signals in the subbands1 through 16 will be normal stereo signals and the signals in thesubbands 17 through 32 will be intensity-stereo signals. Now 48allocation words are needed, which are inserted in the second frameportion, which then has a length of 6 information packets (slots), seeFIG. 22d.

What has been stated above about the scale factors is also valid here.When it is assumed that an allocation word 0000 has been assignedneither to any of the subbands nor to any of the channels, 64 scalefactors are required both for the stereo mode and for theintensity-stereo modes. This is because in all the intensity-stereomodes every mono subband should have two scale factors to enableintensity-stereo to be realised for the left-hand and the right-handchannel in this subband (see FIG. 15c). It is obvious that in the monomode the number of scale factors is halved, i.e. 32, again assuming thatthe allocation word 0000 has not been assigned to any of the subbands.

Scale Factor Determination

A method of determining the 6-bit scale factors will now be explainedbelow. As stated hereinbefore, the sample having the largest absolutevalue is determined for every 12 samples of a subband channel. Line (a)of FIG. 24 shows the binary representation of a maximal sample |S_(max)|. The first bit, designated SGN, is the sign bit and is `0` because itrelates to the absolute value of S_(max). The samples are represented intwo's complement notation. The sample comprises k `zeros` followed by a"1". The values of the other bits of the 24-bit digital number are notrelevant and can be either `0` or `1`.

|S_(max) | is now multiplied by 2^(k) to produce the number shown inline (b) of FIG. 24. Subsequently |S_(max) |0.2^(k) is compared with adigital number DV₁ equal to 010100001100000000000000 and a digitalnumber DV₂ equal to 011001100000000000000000. If |S_(max) |0.2^(k) <DV₁a specific constant p is taken to be 2. If DV₁ ≦|S_(max) |0.2^(k) <DV₂,then p is taken to be 1. If |S_(max) |0.2^(k) ≧DV₂, then p=0.

The number k is limited to 0≦k≦20. The scale factor is now determined bythe numbers k and p in accordance with the following formula.

    SF=3k+p.

Consequently, the maximum value for SF is 62. This means that the scalefactors can be represented by 6-bit numbers, the six-bit number111111(which corresponds to the decimal number 63) not being used. Infact, the 6-bit binary numbers are not the scale factors but they are ina uniquely defined relationship with the actual scale factors, as willbe set forth below. All the 12 samples S are now multiplied by a numberwhich is related to the values for k and p. The 12 samples are eachmultiplied as follows:

    S'=S×2.sup.k ×g(p)

where the number g(p) has the following relation with p:

    g(p)=1 for p=0

    g(p)=1+2.sup.-2 +2.sup.-8 +2.sup.-10 +2.sup.-16 +2.sup.-18 +2.sup.-23 for p=1

    g(p)=1+2.sup.-1 +2.sup.-4 +2.sup.-6 +2.sup.-8 +2.sup.-9 +2.sup.-10 +2.sup.-13 +2.sup.-15 +2.sup.-16 +2.sup.-17 +2.sup.-19 +2.sup.-20 for p=2.

The parameter k specifies the number of 6 dB steps and the factors (g(1)and g(2) are the closest approximations to steps of 2 dB. The samples S'thus scaled are now quantised to enable them to be represented by q-bitdigital numbers in two's complement notation. In FIG. 25 this isillustrated for q=3. The scaled samples S' have values between 30 1 and-1, see FIG. 25a. In the quantiser these samples must be represented byq bits, q corresponding to the allocation value for the relevant subband(channel). Since, as stated above, the q-bit digital number comprisingonly `ones` is not used to represent a sample the total interval from -1to +1 should be divided over 2^(q-1) smaller intervals. For this purposethe scaled samples S' are transformed into the samples S" in accordancewith the formula S"=S'(1-2³¹ q)-2^(-q).

The samples S" are subsequently truncated at q bits, see FIG. 25c. Sincethe `111` representation is not permissible the sign bits are inverted,see FIG. 25d. The q(=3)-bit numbers given in FIG. 25d are now insertedin the third frame portion FD3, see FIG. 2.

Samples S' which comply with -0.71≦S'≦0.14 are represented by thedigital number `001` This proceeds similarly for samples S' of largervalues up to samples which comply with 0.71≦S'<1 and which arerepresented by the digital number "110'. Consequently, the digitalnumber `111` is not used.

Dequantization at the receiving side is effected in a manner inverse tothe quantization at the transmission side, see FIG. 26. This means thatfirst the sign bits of the q-bit digital numbers are inverted to obtainthe normal two's complement notation, see FIG. 26b.

Subsequently the samples S' are derived from the transformed samples S'by means of the formula

    S'=(S"+2.sup.-q+1)(1+2.sup.-q +2.sup.-2q +2.sup.-3q +2.sup.-4q + . . . )

(see FIGS. 26c and 26d). The values S' thus obtained are now situatedexactly within the original intervals in FIG. 25a. At the receiving sidethe samples S' are subsequently scaled to the original amplitudes bymeans of the transmitted information k, p which is related to the scalefactors. Thus, at the receiving side a number g'(p) complies with:

    g'(p)=1 for p=0

    g'(p)=2.sup.-1 +2.sup.-2 +2.sup.-5 +2.sup.-6 for p=1

    g(p)=2.sup.-1 +2.sup.-3 +2.sup.-8 +2.sup.-9 for p=2.

Scaling to the original amplitudes is now effected using the followingformula: S=S'0.2^(-k).g'(p).

In the two possible versions of a frame as described with reference toFIGS. 2 and 3 and FIGS. 2, 17 and 19 respectively the third frameportion may not be filled entirely with information. This will occurmore often and sooner as the algorithms for subband coding, i.e. theentire process of dividing the signal into subband signals and thesubsequent quantization of the samples in the various subbands, areimproved. In particular, this will enable the information to betransmitted with a smaller number of bits (average number per sample).The unused part of the third frame portion can then be utilized fortransmitting additional information. In the first frame portion FD1 inFIG. 17 allowance has been made for this by means of the "future-use"bit b₂₃. Normally, this bit is `0`, as will be apparent from FIG. 18.

Additional Signal

If an additional signal has been inserted in the third frame portion FD3of a frame, the future-use bit b₂₃ in the first frame portion FD1, seeFIG. 17, will be `1`. During reading of the first frame portion FD1 thismakes it possible for the receiver to detect whether the frame containsadditional information. The allocation information and the scalefactors, see FIG. 23, inform the receiver that only the part of thethird frame portion FD3, marked FD4 in FIG. 23, contains quantisedsamples of the subband signals. The remainder, marked FD5 in FIG. 23,now contains the additional information. The first bits in this frameportion FD5 are designated `EXT INFO` or extension information. Thesebits indicate the type of additional information. The additionalinformation may be, for example, an additional audio channel, forexample for the transmission of a second stereo channel. Anotherpossibility is to use these two additional audio channels to realise`surround sound` together with the audio subband signals in the frameportion FD4. In that case the front-rear information required forsurround sound may be included in the frame portion FD5. In the partmarked FD6 the frame portion FD5 may again contain allocationinformation, scale factors and samples (in this order) and the sequenceof the allocation words and the scale factors may then be similar to thesequence as described with reference to FIGS. 2 and 3 and FIGS. 2, 17and 19.

In the case of `surround sound` simple receivers may merely decode thestereo audio information in the frame portions FD2 and FD3, except forthe frame portion FD5. More sophisticated receivers are then capable ofreproducing the surround-sound information and for this purpose theyalso employ the information in the frame portion FD5.

The extension-info bits may also indicate that the information in theframe portion FD6 relates to text, for example in the form of ASCIIcharacters. It may even be considered to insert video or pictureinformation in the frame portion FD6, this information again beingcharacterized by the extension-info bits.

It is to be noted that the invention is not limited to the embodimentsshown herein. The invention also relates to those embodiments whichdiffer from the embodiments shown herein with respect to features whichare not relevant to the invention as defined in the Claims.

What is claimed is:
 1. A digital transmission system, for producing a replica of a digital signal comprising at least a first component and a second component, comprising:an encoder including analysis means for altering said digital signal to obtain a plurality of sub-signals, including at least a first sub-signal and a second sub-signal from said first component and said second component respectively, signal combination means for combining said first sub-signal and said second sub-signal to obtain a composite sub-signal, signal generator means for generating an indicator signal indicating that said first sub-signal and said second sub-signal are combined, transmission means for transmitting said indicator signal, said composite sub-signal, and sub-signals which have not been combined, receiving means for receiving the signals which were transmitted, means, responsive to the received indicator signal and composite subsignal, for generating a signal related to at least one of said first component and said second component, and a decoder including synthesis means for combining the transmitted subsignals and the signal related to at least one of said first component and said second component to produce said replica of the digital signal.
 2. A system as claimed in claim 1, characterized in that said first and second component represent information relating to the stereo nature of an audio signal.
 3. A system as claimed in claim 1, characterized in that said analysis means subdivides said digital signal into sub-signals which are subband signals representing respective frequency subbands and said components.
 4. A digital transmission system, for producing a replica of a digital signal comprising at least a first and a second component, comprising a transmitter and a receiver,wherein said transmitter comprises:an encoder including analysis means for filtering said digital signal to obtain subband signals for M subbands, where M>1, said subband signals including a plurality of first subband signals and a plurality of second subband signals from said first component and second component respectively, signal combination means for combining m₁ of said first subband signals respectively with m₁ of said second subband signals from corresponding subbands to obtain m₁ composite subband signals, where 1<m₁ <M, signal generator means for generating an indicator signal indicating which said first and second subband signals are combined, and transmission means for transmitting said indicator signal, said composite subband signals, and subband signals which were not combined, and the receiver comprises: receiving means for receiving the signals which have been transmitted, detection means for detecting said indicator signal, derivation means responsive to the received indicator signal, for producing, from the received composite subband signals, derived subband signals related to m₁ of said first subband signals and m₁ of said second subband signals, and a decoder including synthesis means for combining said derived subband signals and the received subband signals which were not combined, to produce said replica of the digital signal.
 5. A system as claimed in claim 4, wherein said analysis means applies substantially identical filtering to said first and second components to obtain said first and second subbands.
 6. A system as claimed in claim 4, characterized in that said m₁ of the first subband signals are subband signals for the m₁ highest frequency subbands.
 7. A system as claimed in claim 4, characterized in that said digital signal represents a first block of samples and a second block of samples, said first component and said second component being first block first and second components, said subband signals being first block subband signals, said m₁ of said first and second subband signals being m₁ of the first block first and second subband signals, said m₁ composite subband signals being m₁ first block composite subband signals, and said replica being a replica of the portion of said digital signal representing said first block of samples,for producing a replica of the portion of said digital signals representing said second block of samples, said analysis means obtains second block subband signals for said M subbands including corresponding pluralities of second block first and second subband signals from second block first and second components respectively, said signal combination means combines in each of a number m₂ subbands the second block subband signals from the respective second block first and second components, to obtain m₂ composite signals in said m₂ subbands, where m₂ is greater than m₁, said signal generator means generates a second block indicator signal identifying said m₂ subbands, said transmitter transmits said composite signals in said m₂ subbands, said second block indicator signal, and second block subband signals which were not combined, and said derivation means derives said m₂ composite signals in said m₂ subbands from the second block signal received, and derives from said m₂ composite signals in said m₂ subbands, in response to said second block indicator signal, subband signals in said m₂ subbands corresponding to said second block subband signals which were combined.
 8. A system as claimed in claim 7, characterized in that said m₂ subbands are the m₂ highest subbands.
 9. A system as claimed in claim 6, characterized in that said first and second components are respective stereo audio signals.
 10. A system as claimed in claim 4, characterized in that said transmitter comprises a scale factor determiner, for determining a scale factor for time equivalent subband signal blocks of the first and second components in the subband signals; and means for transmitting these scale factors, andsaid detector in the receiver is adapted to detect the scale factors which have been transmitted.
 11. A system as claimed in claim 2, characterized in that said transmitter comprises means for quantizing the time equivalent signal blocks of the subband signals and the one or more composite signals.
 12. A transmitter, for transmitting signals representative of a digital signal comprising at least a first component and a second component, over a transmission medium, comprising:an encoder including analysis means for altering said digital signal to obtain a plurality of sub-signals, including at least a first sub-signal and a second sub-signal from said first component and said second component respectively, signal combination means for combining said first sub-signal and said second sub-signal to obtain a composite sub-signal, signal generator means for generating an indicator signal indicating that said first sub-signal and said second sub-signal are combined, transmission means for transmitting said indicator signal, said composite sub-signal, and sub-signals which have not been combined.
 13. A transmitter as claimed in claim 12, characterized in that said first and second components are respective stereo audio signals.
 14. A transmitter as claimed in claim 12, characterized in that said analysis means filters said digital signal to provide sub-signals which are subband signals representing said digital signal in M respective frequency subbands, where M>1, said subband signals including a plurality of first subband signals and a plurality of second subband signals from said first component and second component respectively,said signal combination means combines m₁ of said first subband signals respectively with m₁ of said second subband signals from corresponding subbands to obtain m₁ composite subband signals, where 1<m₁ <M, and said indicator signal indicates which said first and second subband signals are combined.
 15. A transmitter as claimed in claim 14, characterized in that said digital signal represents a first block of samples and a second block of samples, said first component and said second component being first block first and second components, said subband signals being first block subband signals, said m₁ of said first and second subband signals being m₁ of the first block first and second subband signals, said m₁ composite subband signals being m₁ first block composite subband signals, and said replica being a replica of the portion of said digital signal representing said first block of samples,for producing a replica of the portion of said digital signals representing said second block of samples, said analysis means obtains second block subband signals for said M subbands including corresponding pluralities of second block first and second subband signals from second block first and second components respectively, said signal combination means combines in each of a number m₂ subbands the second block subband signals from the respective second block first and second components, to obtain m₂ composite signals in said m₂ subbands, where m₁ <m₂ ≦M, said signal generator means generates a second block indicator signal identifying said m₂ subbands, and said transmission means transmits said m₂ composite signals, said second block indicator signal, and second block subband signals which were not combined.
 16. A transmitter as claimed in claim 14, wherein said analysis means applies substantially identical filtering to said first and second components to obtain said first and second subbands.
 17. A transmitter as claimed in claim 14, characterized in that said m₁ of the first subband signals are subband signals for the m₁ highest frequency subbands.
 18. A transmitter as claimed in claim 14, characterized in that said transmitter comprises a scale factor determiner, for determining a scale factor for time equivalent subband signal blocks of the first and second components in the subband signals; and means for transmitting these scale factors.
 19. A receiver for producing a replica of a digital signal including a first component and a second component, from digital signal components comprising at least one composite sub-signal, an indicator signal indicating that at least a first and a second sub-signal are combined, and a plurality of subsignals not including said first and second sub-signal, said digital signal components being representative of said digital signal,receiving means for receiving said digital signal components, means, responsive to the received indicator signal and composite subsignal, for generating a signal related to at least one of said first component and said second component, and a decoder including synthesis means for combining the transmitted subsignals and the signal related to at least one of said first component and said second component to produce said replica of the digital signal.
 20. A receiver as claimed in claim 19, characterized in that said first and second components are respective stereo audio signals.
 21. A receiver as claimed in claim 19, characterized in that said composite sub-signal represents a first subband signal and a second subband signal for a combined subband, and said sub-signals are subband signals representing respective frequency subbands other than said combined subband, andsaid means for generating comprises derivation means for deriving said composite subsignal from the signal received and for deriving from said composite signal, in response to the indicator signal, subband signals corresponding to said first component and said second component.
 22. A receiver as claimed in claim 21, characterized in that the digital signal components comprise a plurality of said composite sub-signals representing a plurality of combined subbands respectively, and a scale factor for time equivalent signal blocks of the first component and the second component,said detector in the receiver is adapted to detect said scale factor, and said derivation means is responsive to said scale factor. 